In a communication system a communication network is provided, which can link together two communication terminals so that the terminals can send information to each other in a call or other communication event. Information may include speech, text, images or video.
Modern communication systems are based on the transmission of digital signals. Analogue information such as speech is input into an analogue to digital converter at the transmitter of one terminal and converted into a digital signal. The digital signal is then encoded and placed in data packets for transmission over a channel to the receiver of another terminal.
One type of communication network suitable for transmitting data packets is the internet. Protocols which are used to carry voice signals over an Internet Protocol network are commonly referred to as Voice over IP (VoIP). VoIP is the routing of voice conversations over the Internet or through any other IP-based network.
A data packet includes a header portion and a payload portion. The header portion of the data packet contains data for transmitting and processing the data packet. This information may include an identification number and source address that uniquely identifies the packet, a header checksum used to detect processing errors and the destination address. The payload portion of the data packet includes information from the digital signal intended for transmission. This information may be included in the payload as encoded frames such as voice frames, wherein each frame represents a portion of the analogue signal.
Degradations in the channel on which the information is sent will affect the information received at the receiving terminal. Degradations in the channel can cause changes in the packet sequence, delay the arrival of some packets at the receiver and cause the loss of other packets. The degradations may be caused by channel imperfections, noise and overload in the channel. This ultimately results in a reduction of the quality of the signal output by the receiving terminal.
When data packets are received at the destination terminal, the information provided in the header of each packet is used to order the received data packets in the correct sequence. In order to ensure that the data in the data packets may be output continuously at the destination terminal, it is necessary to introduce a delay between receiving a data packet and outputting the data in the packet, in order to overcome random variations in the delay between packets arriving at the terminal.
In order to ensure that the data in the data packets may be output continuously at the destination terminal, it is necessary to introduce a delay between receiving a data packet and outputting the data in the packet, in order to over come random variations in the in the delay between packets arriving at the terminal.
A jitter buffer is used at the receiving terminal to introduce a delay between receiving data packets from the network and outputting the data from the terminal. The jitter buffer stores packets temporarily to buffer the variations in the arrival times of packets, such that a decoder can take frames out of the jitter buffer on a continuous basis.
A jitter buffer manager is arranged to control the amount of frames in the jitter buffer over time. The jitter buffer manager may control the number of frames in the jitter buffer, thereby adjusting the delay introduced by the jitter buffer, by requesting that the decoder performs an action that will affect the time that the decoder requires the next frame from the jitter buffer.
In order to delay the time that the decoder requires the next frame, the jitter buffer manager is arranged to request that the decoder inserts a copy of the last frame or extents the play out time of a frame, for example by extending the length of the frame by 20 ms to 30 ms. Conversely in order to reduce the time that the decoder requires the next frame, the jitter buffer manager is arranged to request that the decoder skips a frame or shortens the play out time of a frame. If however the delay introduced by the jitter buffer does not need to be altered the jitter buffer manager may request that the decoder decodes the frame.
Simple jitter buffers introduce a delay by storing a predetermined number of packets, before outputting the data in the packets. However it is advantageous to adapt the number of packets stored in the buffer to effectively handle changing network conditions. Therefore, in some methods known in the art, the number of frames to be stored in the jitter buffer may be calculated adaptively.
The longer the delay introduced by the jitter buffer, the less the risk that a packet will be delayed beyond the time at which it is needed; on the other hand, a high delay tends to disrupt two way communication. Therefore, it is crucial to determine a reasonable jitter buffer delay.
Usually, the jitter buffer delay is controlled by estimating the amount of network delay for each packet, and then smoothing these values over time. Alternatively, instead of smoothing, a histogram of the packet delays can be kept at the receiver. A target jitter buffer delay may then be chosen to correspond to the most frequently observed delay. However, controlling the adaptation of the delay in this manner does not allow the jitter buffer delay to adapt sufficiently quickly when the conditions of the network change rapidly.
It is therefore an aim of the present invention to improve the perceived quality of the received signal. It is a further aim of the present invention to provide a method of improving the quality of the received signal without the use of complex computational methods.